similar to: Updated Vorbis-RTP Internet Draft

Hi all, Below is the Vorbis RTP Internet Draft as sent to the AVT working group of the IETF. Comments and feedback is still welcomed from the Vorbis community. Cheers Phil ---------------------------8<-----------------8<------------------------ Network Working Group Phil Kerr Internet-Draft The Ogg Vorbis January 07, 2003 Community / OpenDrama

Hi all, Apologies in advance, this email is quite long. I've prepared an updated Vorbis RTP Internet Draft, which is a continuation of draft-moffitt-vorbis-rtp-00.txt which can be found below. If this new draft gets the ok I'd like to submit this to the AVT WG later this week. There are a number of changes over the original I-D, notably the changing of the M bit function in the RTP

Hi everybody, this is an update on the developed Ogg IETF standard documents and their status. All of these documents are in the process of discussion and have not yet been accepted as standards. <p>The following Internet-Drafts (I-Ds) have been prepared for standardisation and submitted to the IETF: 1) an I-D requesting to register "application/ogg" as a mime-type written by

Hi everybody, this is an update on the developed Ogg IETF standard documents and their status. All of these documents are in the process of discussion and have not yet been accepted as standards. <p>The following Internet-Drafts (I-Ds) have been prepared for standardisation and submitted to the IETF: 1) an I-D requesting to register "application/ogg" as a mime-type written by

Hi everybody, this is an update on the developed Ogg IETF standard documents and their status. All of these documents are in the process of discussion and have not yet been accepted as standards. <p>The following Internet-Drafts (I-Ds) have been prepared for standardisation and submitted to the IETF: 1) an I-D requesting to register "application/ogg" as a mime-type written by

Hi all, I'm in the final stages of putting a new Vorbis RTP draft together, the new sections are below. There are a number of open questions: 1.) At present there is a 16 bit length field for the codebooks. As they are spec'd as being unbound in length, but typically around 15K, are there situations where they may be greater than 64KB? The size limit can be extended to 2MB as there

Hi all, Please find below the -01 update to draft-herlein-speex-rtp-profile, as submitted to the IETF. Regards Phil <p>-------------------8<-----------------------------------8<--------------------- <p><p>Internet Engineering Task Force Greg Herlein Internet Draft Jean-Marc Valin

Hello, What's the purpose of the 'sr' sdp parameter ? The sample rate is already given in the a=rtpmap line ? Simon Le dim 29/06/2003 à 12:12, philkerr@elec.gla.ac.uk a écrit : > Hi all, > > Please find below an updated Speex Internet Draft document. > > It would be good if we could book some time for discussion on Speex at the IETF > meeting in Vienna (scheduled

Hi all, Please find below an updated Speex Internet Draft document. It would be good if we could book some time for discussion on Speex at the IETF meeting in Vienna (scheduled for 14th July). The cutoff for submission is 9:00am EDT, (GMT -04:00), 30th June. Comments and feedback welcomed! Regards Phil

Hi all, Please find below an I-D for the proposed registration of MIME type audio/rtp-vorbis. All comments most welcome. Barry <p>-----------8<------------- Network Working Group Barry Short Internet-Draft The Ogg Vorbis Community / OpenDrama Expires: July 7, 2003 January 7, 2003 <p>

Hi, A radio is being developped at school, and we wanted to avoid eating too much bandwidth, so we had a look at RTP. I first implemented it in icecast1 and it worked well (although we had to apply a patch to make RTP work with xmms). We are indeed saving a lot of bandwidth. I then adapted it to icecast2, here is the patch. The biggest trouble for now is the configurability. I added 3 global

(hello all, this is my first writing. so please bear with me if I'm wrong anywhere.) orry to break too lately, but how is the RTP payload submission is going? could we see the new payload at March IETF? I agree that it would be fairy straightforward to make an RTP payload for ogg vorbis, assuming raw packets, AFAIK. using physical bitstream is, in this case, not adequate by the reasons in

Good day, FYI, I've just sent the attached email to IETF to progress our I-Ds. There are also links to updated documents in this email. Cheers, Silvia. -------- Original Message -------- Subject: New I-Ds for Ogg technologies (Vorbis over RTP, Ogg file format, Mimetypes) Date: Sun, 12 Jan 2003 01:36:31 +1100 From: Silvia Pfeiffer <Silvia.Pfeiffer@csiro.au> Organization: CSIRO To: AVT

Good day, FYI, I've just sent the attached email to IETF to progress our I-Ds. There are also links to updated documents in this email. Cheers, Silvia. -------- Original Message -------- Subject: New I-Ds for Ogg technologies (Vorbis over RTP, Ogg file format, Mimetypes) Date: Sun, 12 Jan 2003 01:36:31 +1100 From: Silvia Pfeiffer <Silvia.Pfeiffer@csiro.au> Organization: CSIRO To: AVT

Alfred E. Heggestad wrote: > <...> > > If we don't get any comments in 1 week (by 22. May 2007) we will go ahead > and submit it to the IETF. Of course you can comment on it also after it > has been submitted, but we would like to get the input from the Speex > community first.. > thanks for all the input. please find attached an updated version of the draft. I

Hi Please find an updated version of the Speex I-D attached. The only change is addition of the copyright conditions in Appendix A, as requested by Ivo. Many thanks for your input. I will give you a few more days before submitting to AVT working group /alfred Ivo Emanuel Gon?alves wrote: > Do not forget to add the "Copying conditions" to the RFC. > > Check

Hello, The Asterisk server is configured to accept calls through h323 net. When calls is came from cisco pstn gate then DTMF tone not recognized, but when from other h323 endpoint then all work fine. I thought the cisco does't send DTMF event but in rtp dump I saw strange packet where DTMF should be: RTP Packet: Size: 36 Version: 2 Padding: FALSE Extension: FALSE marker: FALSE

hi, is there someone with working asterisk13+chan_sip+SIP.js/sipml5 ? or is chan_pjsip better supported? or the recommended way for asterisk is use respoke.io? my problem with asterisk13+chan_sip+sipml5(the same problem is with SIP.js) chan_sip.c:10496 process_sdp: Can't provide secure audio requested in SDP offer " sip.conf (important parts) [vr1a882] ... nat=force_rport,comedia

Looks good to me. Jean-Marc Alfred E. Heggestad a ?crit : > Hi > > Please find an updated version of the Speex I-D attached. The only > change is addition of the copyright conditions in Appendix A, > as requested by Ivo. > > Many thanks for your input. > > I will give you a few more days before submitting to AVT working group > > > /alfred > > Ivo

All: The latest draft RTP Payload Format for Speex is available via the IETF. See below for details. Greg ---------- Forwarded message ---------- Date: Tue, 09 Mar 2004 15:56:23 -0500 From: Internet-Drafts@ietf.org To: IETF-Announce: ; Subject: I-D ACTION:draft-herlein-avt-rtp-speex-00.txt A New Internet-Draft is available from the on-line Internet-Drafts directories. <p>